view libao2/ao_dxr2.c @ 18150:710d4bc5f8c9

Using channel count, samplerate and input bps values from the container instead of the decoder breaks some DTS samples where the container says the audio has 6 channels but the decoder gives 2. In this case take the number of channels from the decoder instead, the output will almost certainly be badly garbled anyway if the number of channels is wrong. patch by Uoti Urpala, uoti <<.>> urpala <<@>> pp1 <<.>> inet <<.>> fi

#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <inttypes.h>
#include <dxr2ioctl.h>
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "bswap.h"

#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"


static ao_info_t info =
{
	"DXR2 audio output",
	"dxr2",
	"Tobias Diedrich <[e-mail]>",
	""
};

LIBAO_EXTERN(dxr2)

static int volume=19;
extern int dxr2_fd;

// to set/get/query special features/parameters
static int control(int cmd,void *arg){
  switch(cmd){
  case AOCONTROL_GET_VOLUME:
    if(dxr2_fd > 0) {
      ao_control_vol_t* vol = (ao_control_vol_t*)arg;
      vol->left = vol->right = volume * 19.0 / 100.0;
      return CONTROL_OK;
    }
    return CONTROL_ERROR;
  case AOCONTROL_SET_VOLUME:
    if(dxr2_fd > 0) {
      dxr2_oneArg_t v;
      float diff;
      ao_control_vol_t* vol = (ao_control_vol_t*)arg;
      // We need this trick because the volume stepping is often too small
      diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
      v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff)); 
      if(v.arg > 19) v.arg = 19;
      if(v.arg < 0) v.arg = 0;
      if(v.arg != volume) {
	volume = v.arg;
	if( ioctl(dxr2_fd,DXR2_IOC_SET_AUDIO_VOLUME,&v) < 0) {
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_SetVolFailed,volume);
	  return CONTROL_ERROR;
	}
      }
      return CONTROL_OK;
    }
    return CONTROL_ERROR;
  }
  return CONTROL_UNKNOWN;
}

static int freq=0;
static int freq_id=0;

// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){

	if(dxr2_fd <= 0)
	  return 0;

	ao_data.outburst=2048;
	ao_data.samplerate=rate;
	ao_data.channels=channels;
	ao_data.buffersize=2048;
	ao_data.bps=rate*4;
	ao_data.format=format;
	freq=rate;

	switch(rate){
	case 48000:
		freq_id=DXR2_AUDIO_FREQ_48;
		break;
	case 96000:
		freq_id=DXR2_AUDIO_FREQ_96;
		break;
	case 44100:
		freq_id=DXR2_AUDIO_FREQ_441;
		break;
	case 32000:
		freq_id=DXR2_AUDIO_FREQ_32;
		break;
	case 22050:
		freq_id=DXR2_AUDIO_FREQ_2205;
		break;
#ifdef DXR2_AUDIO_FREQ_24
	// This is not yet in the dxr2 driver CVS
	// you can get the patch at
	// http://www.tdiedrich.de/~ranma/patches/dxr2.pcm1723.20020513
	case 24000:
		freq_id=DXR2_AUDIO_FREQ_24;
		break;
	case 64000:
		freq_id=DXR2_AUDIO_FREQ_64;
		break;
	case 88200:
		freq_id=DXR2_AUDIO_FREQ_882;
		break;
#endif
	default:
		mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_DXR2_UnsupSamplerate,rate);
		return 0;
	}

	return 1;
}

// close audio device
static void uninit(int immed){

}

// stop playing and empty buffers (for seeking/pause)
static void reset(){

}

// stop playing, keep buffers (for pause)
static void audio_pause()
{
    // for now, just call reset();
    reset();
}

// resume playing, after audio_pause()
static void audio_resume()
{
}

extern void dxr2_send_packet(unsigned char* data,int len,int id,int timestamp);
extern void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,int timestamp,int freq_id);
extern int vo_pts;
// return: how many bytes can be played without blocking
static int get_space(){
    float x=(float)(vo_pts-ao_data.pts)/90000.0;
    int y;
    if(x<=0) return 0;
    y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
    if(y>32768) y=32768;
    return y;
}

// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
  // MPEG and AC3 don't work :-(
    if(ao_data.format==AF_FORMAT_MPEG2)
	dxr2_send_packet(data,len,0xC0,ao_data.pts);
    else if(ao_data.format==AF_FORMAT_AC3)
      	dxr2_send_packet(data,len,0x80,ao_data.pts);
    else {
	int i;
	//unsigned short *s=data;
	uint16_t *s=data;
#ifndef WORDS_BIGENDIAN
	for(i=0;i<len/2;i++) s[i] = bswap_16(s[i]);
#endif
	dxr2_send_lpcm_packet(data,len,0xA0,ao_data.pts-10000,freq_id);
    }
    return len;
}

// return: delay in seconds between first and last sample in buffer
static float get_delay(){

    return 0.0;
}